Merge master and multifarious hackery.

[dcpomatic.git] / src / lib / audio_decoder.cc

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index a54c14843927449b4a62f00665ea83020a53bf70..9b8d15bf163927c69cbe201e5c7427825013db76 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -18,19 +18,140 @@
 */
 
 #include "audio_decoder.h"
-#include "stream.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+#include "log.h"
 
+#include "i18n.h"
+
+using std::stringstream;
+using std::list;
+using std::pair;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoder::AudioDecoder (shared_ptr<Film> f, DecodeOptions o)
-       : Decoder (f, o)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
+       : Decoder (f)
+       , _next_audio (0)
+       , _audio_content (c)
 {
+       if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) {
+
+               shared_ptr<const Film> film = _film.lock ();
+               assert (film);
+
+               stringstream s;
+               s << String::compose (
+                       "Will resample audio from %1 to %2",
+                       _audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate()
+                       );
+               
+               film->log()->log (s.str ());
+
+               /* We will be using planar float data when we call the
+                  resampler.  As far as I can see, the audio channel
+                  layout is not necessary for our purposes; it seems
+                  only to be used get the number of channels and
+                  decide if rematrixing is needed.  It won't be, since
+                  input and output layouts are the same.
+               */
+
+               _swr_context = swr_alloc_set_opts (
+                       0,
+                       av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+                       AV_SAMPLE_FMT_FLTP,
+                       _audio_content->output_audio_frame_rate(),
+                       av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+                       AV_SAMPLE_FMT_FLTP,
+                       _audio_content->content_audio_frame_rate(),
+                       0, 0
+                       );
+               
+               swr_init (_swr_context);
+       } else {
+               _swr_context = 0;
+       }
+}
 
+AudioDecoder::~AudioDecoder ()
+{
+       if (_swr_context) {
+               swr_free (&_swr_context);
+       }
 }
+       
 
+#if 0
 void
-AudioDecoder::set_audio_stream (shared_ptr<AudioStream> s)
+AudioDecoder::process_end ()
 {
-       _audio_stream = s;
+       if (_swr_context) {
+
+               shared_ptr<const Film> film = _film.lock ();
+               assert (film);
+               
+               shared_ptr<AudioBuffers> out (new AudioBuffers (film->audio_mapping().dcp_channels(), 256));
+                       
+               while (1) {
+                       int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
+
+                       if (frames < 0) {
+                               throw EncodeError (_("could not run sample-rate converter"));
+                       }
+
+                       if (frames == 0) {
+                               break;
+                       }
+
+                       out->set_frames (frames);
+                       _writer->write (out);
+               }
+
+       }
 }
+#endif
+
+void
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
+{
+       /* Maybe resample */
+       if (_swr_context) {
+
+               /* Compute the resampled frames count and add 32 for luck */
+               int const max_resampled_frames = ceil (
+                       (int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate()
+                       ) + 32;
+
+               shared_ptr<AudioBuffers> resampled (new AudioBuffers (data->channels(), max_resampled_frames));
+
+               /* Resample audio */
+               int const resampled_frames = swr_convert (
+                       _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames()
+                       );
+               
+               if (resampled_frames < 0) {
+                       throw EncodeError (_("could not run sample-rate converter"));
+               }
+
+               resampled->set_frames (resampled_frames);
+               
+               /* And point our variables at the resampled audio */
+               data = resampled;
+       }
+
+       shared_ptr<const Film> film = _film.lock ();
+       assert (film);
+       
+       /* Remap channels */
+       shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
+       dcp_mapped->make_silent ();
+       list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
+       for (list<pair<int, libdcp::Channel> >::iterator i = map.begin(); i != map.end(); ++i) {
+               dcp_mapped->accumulate_channel (data.get(), i->first, i->second);
+       }
+
+       Audio (dcp_mapped, time);
+       _next_audio = time + film->audio_frames_to_time (data->frames());
+}
+
+